Displaying 20 results from an estimated 20000 matches similar to: "Asterisk, Configuration of SDP in SIP messages"
2005 Jun 26
1
help regarding h323.conf
2013 Jun 28
1
Asterisk behind NAT and Kamailio --> Internal IP in SDP and not "externip"
Hi,
We have some Asterisk servers that we are moving behind a NAT to
preserve public addresses and make room for growth. This is Asterisk 1.4
NAT works very good with the externip/localnet-setting when we are
connected directly to our teleco. But when I try to use NAT and put them
behind our Kamailio something interesting happens: The media-address in
the SDP is the internal ip and not the
2014 Jun 18
1
Making sense of SDP for debugging of missing audio in SIP trunk
I am debugging an intermittent issue of missing audio on calls that come from a SIP provider into our asterisk-11.10 installation. Sometimes, incoming calls from this provider work correctly, with audio streaming in both directions. Other times, with the
same setup, the calling party is unable to hear the IVR recording from the asterisk installation, although in fact the streaming is supposed to
2009 Nov 24
1
Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = ' ')
Hi,
I am using codec g729 on two asterisk machines, but when call is forwarded
from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs
following error and there is no audio. Also the IVRs being played have
choppy voice.
"Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')"
It is running fine when codec gsm is in RTP traffic.
Also I
2003 Jun 10
1
SIP sdp o= and c= fields
Hello,
If I understand it correctly, when sending INVITE, o= and c= sdp fields are
built using p->ourip
IP address. At this point RTP packets will be coming to the default asterisk
IP address.
For the machine with multiple interfaces this could be not the right one
(not what we want).
Could it be configured (in rtp.conf or in sip.conf per context) ?
Thank you.
Alex Zarubin
--------------
2023 Jun 28
1
SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
Hello list
when trying to set up webRTC communications with sipjs client package
(tried 0.7.0, 0.10.0 and 0.19.0), I see in the asterisk debug log-file
the following :
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
c=IN IP4 99.88.77.66... OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=rtcp:9 IN IP4 0.0.0.0... UNSUPPORTED OR FAILED.
2009 Sep 04
1
Send 200 OK with SDP instead of 183 with SDP when ringing starts
Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through
which I connected an external PSTN line. I use it as carrier for VoIP
calls. I can make successfully calls, but there's one problem, I receive
200 OK with SDP with delay (sometimes more than 30 seconds).
So when I make a call through asterisk I receive intially:
- 100 Trying
- 183 Session Progress, with SDP
when the called
2014 May 30
1
Configuring Asterisk to allow any payload type in SDP
Hello,
Is there a way to configure Asterisk so that it doesn't care about the
playload type in SDP ?
I'm trying to send custom data which has assigned a dynamic PT through RTP
so I only need Asterisk to act as a proxy/forwarder but I'm getting "488
Not acceptable here" responses and in console "No compatible codecs, not
accepting this offer"
Best regards,
Nicolae
2012 Jan 28
1
process_sdp: Unsupported SDP media type in offer: audio , Failing due to no acceptable offer found
Hi All,
I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
But when making A Call from SIP Client, I got cli Warning ... and no call
has been made.
My Sip Client is using lib java peers client http://peers.sourceforge.net/
with standard codec PCMU/PCMA
[Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp: Unsupported
SDP media type in offer: audio 0 RTP/AVP 0 8
2017 Oct 06
2
Asterisk put call on hold when receive 183 Session Progress with media address 0.0.0.0 in SDP
Hi
Is it a normal behavior of Asterisk put a call on hold when receive a
Session Progress with media address 0.0.0.0 in SDP? I believe the call on
hold should be initiate with a re-invite.
Thanks
--
Att,
Rafael Saraiva
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2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
Hello,
I have a problem with a call between 2 webrtc clients. Asterisk removes the
ice-related lines from the sdp when it sends the INVITE out, and the called
webrtc client rejects the INVITE due to the missing ice lines. Both webrtc
clients are defined exactly the same way, same values in all fields except
the number of the peer.
There's probably something I've changed that causes this
2002 Dec 10
5
Using the right network interface
Hi everyone,
samba 2.2.5
The server I'm using has 2 interfaces so using the
interface parameter I'm telling samba to use eth0 but
for some reason when I do netstat it is listening on eth1
interface = eth0 (the IP is 192.168.6.10)
netstat -an
udp 0 0 138.79.161.225:137 0.0.0.0:*
udp 0 0 0.0.0.0:137 0.0.0.0:*
udp 0 0 138.79.161.225:138
2006 Nov 22
0
Is it easy to route SIP/SDP and SIP/RTP through different routes ?
Hello,
The setup is :
Asterisk ------<www> ------- Router1---------<LAN> --------- SIP Phones
| |
|------------------<pstn>----------Router2------------|
(I hope the sketch is understandable. If not, it could be summarized with :
"SIP hardphones on a given LAN are connected to an Asterisk server through 2
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
Hello,
I was testing with sdp and something came up worth asking:
While calling from a webrtc client to another (chrome, sip.js) Asterisk
receives the following sdp and rejects it with 488 Not Acceptable. Why does
this happen, what's wrong with the sdp? The second sdp body below is
accepted instead. Both have rtp profile RTP/SAVPF, difference is that the
second one was produced by rtpengine,
2009 May 27
2
problem with T.38 media headers
Hi Guys,
Something I have noticed while dealing with T.38 and re-invites in Asterisk 1.4.22.
I have a provider who re-invites with the following sdp (message flow
PROVIDER_EQPMT -> ASTERISK):
"""
.
v=0.
o=SIP_5F9 123456 654322 IN IP4 CONN_IP_PROVIDER.
s=-.
c=IN IP4 CONN_IP_PROVIDER.
t=0 0.
m=audio 0 RTP/AVP 0.
m=image 26858 udptl t38.
a=T38FaxMaxBuffer:288.
2007 Feb 03
2
Powerware 5110
I have an FC5 system with a powerware 5110 usb ups.
The system seems to find the UPS
------------------------------
[root@b1 src]# lsusb
Bus 004 Device 001: ID 0000:0000
Bus 003 Device 002: ID 0592:0002 Powerware Corp.
Bus 003 Device 001: ID 0000:0000
Bus 002 Device 001: ID 0000:0000
Bus 005 Device 001: ID 0000:0000
Bus 001 Device 001: ID 0000:0000
-------------------------------
[root@b1
2004 Jan 22
3
R2 or E&M for E1 CAS pbx to pbx link
2007 Sep 07
0
Errors: Too many SIP headers and Unknown SDP media type in offer: video 10702 RTP/AVP 34 31
Dear all,
I have Asterisk 1.2.13 running OK with Twinkle clients, they can talk
very well using SIP.
I have a Jabber server running OK and the clients use PSI client for
chat succesfully.
Now I'm using Wengophone 2.1.1 in order to unify voip+IM services. The
users can logon OK in SIP and Jabber, they get the online status
presence, but they CAN'T talk and chat among them.
2007 Jun 18
1
180 Ringing with SDP
We're dialing a disconnected number via Level 3's vector network, and
are receiving this. The response has SDP in it. Apparently, Level 3 is
playing early media. Asterisk doesn't seem to know what to do with SDP
in a 180 RINGING, and just plays ringing. What am I missing here? How
can Asterisk see there's SDP, early media, in the response and act
accordingly?
SIP/2.0 180
2014 Apr 09
2
I can't make outbound calls (status is 'CHANUNAVAIL')
Hello:
I have this situation: I can make calls internally, I can make inbound
calls but I can't make outbound calls.
Thanks in advance.
These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
port 1 (FXS) connected to an analog phone
port 3 (FXO) connected to the PSTN
These are my